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fteqw/engine/client/sound.h
Spoike df22ebd757 Tweaks for voicechat:
Added cl_voip_test cvar, so you can hear/test yourself.
Added potential support for opus.
audio ducking
autogain switched off by default. its nice for pure speech but shite for push-to-talk, voice activation, or indeed anything else.

git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@4375 fc73d0e0-1445-4013-8a0c-d673dee63da5
2013-05-31 01:16:07 +00:00

324 lines
11 KiB
C

/*
Copyright (C) 1996-1997 Id Software, Inc.
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
See the GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
// sound.h -- client sound i/o functions
#ifndef __SOUND__
#define __SOUND__
// !!! if this is changed, it much be changed in asm_i386.h too !!!
#define MAXSOUNDCHANNELS 8 //on a per device basis
// !!! if this is changed, it much be changed in asm_i386.h too !!!
struct sfx_s;
/*typedef struct
{
int left;
int right;
} portable_samplepair_t;
*/
typedef struct
{
int s[MAXSOUNDCHANNELS];
} portable_samplegroup_t;
typedef struct {
struct sfxcache_s *(*decodedata) (struct sfx_s *sfx, struct sfxcache_s *buf, int start, int length); //retrurn true when done.
void (*abort) (struct sfx_s *sfx); //it's not playing elsewhere. free entirly
void *buf;
} sfxdecode_t;
typedef struct sfx_s
{
char name[MAX_OSPATH];
#ifdef AVAIL_OPENAL
unsigned int openal_buffer;
#endif
qboolean failedload:1; //no more super-spammy
qboolean touched:1; //if the sound is still relevent
sfxdecode_t decoder;
} sfx_t;
// !!! if this is changed, it much be changed in asm_i386.h too !!!
typedef struct sfxcache_s
{
unsigned int length; //sample count
unsigned int loopstart; //-1 or sample index to begin looping at once the sample ends
unsigned int speed;
unsigned int width;
unsigned int numchannels;
unsigned int soundoffset; //byte index into the sound
qbyte *data; // variable sized
} sfxcache_t;
typedef struct
{
// qboolean gamealive;
// qboolean soundalive;
// qboolean splitbuffer;
int numchannels; // this many samples per frame
int samples; // mono samples in buffer (individual, non grouped)
// int submission_chunk; // don't mix less than this #
int samplepos; // in mono samples
int samplebits;
int speed; // this many frames per second
unsigned char *buffer; // pointer to mixed pcm buffer (not directly used by mixer)
} dma_t;
#define PITCHSHIFT 6 /*max audio file length = (1<<32>>PITCHSHIFT)/KHZ*/
#define CF_ABSVOLUME 1 // ignores volume cvar.
typedef struct
{
sfx_t *sfx; // sfx number
int vol[MAXSOUNDCHANNELS]; // volume, .8 fixed point.
int start; // start time in global paintsamples
int pos; // sample position in sfx, <0 means delay sound start (shifted up by 8)
int rate; // 24.8 fixed point rate scaling
int flags; // cf_ flags
int looping; // where to loop, -1 = no looping
int entnum; // to allow overriding a specific sound
int entchannel; //int audio_fd
vec3_t origin; // origin of sound effect
vec_t dist_mult; // distance multiplier (attenuation/clipK)
int master_vol; // 0-255 master volume
} channel_t;
typedef struct
{
int rate;
int width;
int numchannels;
int loopstart;
int samples;
int dataofs; // chunk starts this many bytes from file start
} wavinfo_t;
struct soundcardinfo_s;
typedef struct soundcardinfo_s soundcardinfo_t;
void S_Init (void);
void S_Startup (void);
void S_Shutdown (void);
void S_StartSound (int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float fvol, float attenuation, float timeofs, float pitchadj);
void S_StaticSound (sfx_t *sfx, vec3_t origin, float vol, float attenuation);
void S_StopSound (int entnum, int entchannel);
void S_StopAllSounds(qboolean clear);
void S_UpdateListener(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up);
void S_GetListenerInfo(float *origin, float *forward, float *right, float *up);
void S_Update (void);
void S_ExtraUpdate (void);
void S_MixerThread(soundcardinfo_t *sc);
void S_Purge(qboolean retaintouched);
qboolean S_HaveOutput(void);
void S_Music_Clear(sfx_t *onlyifsample);
void S_Music_Seek(float time);
sfx_t *S_PrecacheSound (char *sample);
void S_TouchSound (char *sample);
void S_UntouchAll(void);
void S_ClearPrecache (void);
void S_BeginPrecaching (void);
void S_EndPrecaching (void);
void S_PaintChannels(soundcardinfo_t *sc, int endtime);
void S_InitPaintChannels (soundcardinfo_t *sc);
void S_ShutdownCard (soundcardinfo_t *sc);
void S_DefaultSpeakerConfiguration(soundcardinfo_t *sc);
void S_ResetFailedLoad(void);
#ifdef PEXT2_VOICECHAT
void S_Voip_Parse(void);
#endif
#ifdef VOICECHAT
extern cvar_t cl_voip_showmeter;
void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf);
void S_Voip_MapChange(void);
int S_Voip_Loudness(qboolean ignorevad); //-1 for not capturing, otherwise between 0 and 100
qboolean S_Voip_Speaking(unsigned int plno);
void S_Voip_Ignore(unsigned int plno, qboolean ignore);
#else
#define S_Voip_Loudness() -1
#define S_Voip_Speaking(p) false
#define S_Voip_Ignore(p,s)
#endif
qboolean S_IsPlayingSomewhere(sfx_t *s);
qboolean ResampleSfx (sfx_t *sfx, int inrate, int inchannels, int inwidth, int insamps, int inloopstart, qbyte *data);
// picks a channel based on priorities, empty slots, number of channels
channel_t *SND_PickChannel(soundcardinfo_t *sc, int entnum, int entchannel);
// spatializes a channel
void SND_Spatialize(soundcardinfo_t *sc, channel_t *ch);
void SND_ResampleStream (void *in, int inrate, int inwidth, int inchannels, int insamps, void *out, int outrate, int outwidth, int outchannels, int resampstyle);
// restart entire sound subsystem (doesn't flush old sounds, so make sure that happens)
void S_DoRestart (void);
void S_SetUnderWater(qboolean underwater);
void S_Restart_f (void);
//plays streaming audio
void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels, int width, float volume);
void CLVC_Poll (void);
void SNDVC_MicInput(qbyte *buffer, int samples, int freq, int width);
#ifdef AVAIL_OPENAL
void OpenAL_LoadCache(sfx_t *s, sfxcache_t *sc);
void OpenAL_StartSound(int entnum, int entchannel, sfx_t * sfx, vec3_t origin, float fvol, float attenuation, float pitchscale);
void OpenAL_Update_Listener(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up, vec3_t velocity);
void OpenAL_CvarInit(void);
#endif
// ====================================================================
// User-setable variables
// ====================================================================
#define MAX_CHANNELS 1024/*tracked sounds (including statics)*/
#define MAX_DYNAMIC_CHANNELS 64 /*playing sounds (identical ones merge)*/
#define NUM_MUSICS 1
#define AMBIENT_FIRST 0
#define AMBIENT_STOP NUM_AMBIENTS
#define MUSIC_FIRST AMBIENT_STOP
#define MUSIC_STOP (MUSIC_FIRST + NUM_MUSICS)
#define DYNAMIC_FIRST MUSIC_STOP
#define DYNAMIC_STOP (DYNAMIC_FIRST + MAX_DYNAMIC_CHANNELS)
//
// Fake dma is a synchronous faking of the DMA progress used for
// isolating performance in the renderer. The fakedma_updates is
// number of times S_Update() is called per second.
//
extern int snd_speed;
extern vec3_t listener_origin;
extern vec3_t listener_forward;
extern vec3_t listener_right;
extern vec3_t listener_up;
extern vec_t sound_nominal_clip_dist;
extern cvar_t loadas8bit;
extern cvar_t bgmvolume;
extern cvar_t volume;
extern cvar_t snd_capture;
extern float voicevolumemod;
extern qboolean snd_initialized;
extern cvar_t snd_usemultipledevices;
extern cvar_t snd_mixerthread;
extern int snd_blocked;
void S_LocalSound (char *s);
qboolean S_LoadSound (sfx_t *s);
typedef qboolean (*S_LoadSound_t) (sfx_t *s, qbyte *data, int datalen, int sndspeed);
qboolean S_RegisterSoundInputPlugin(S_LoadSound_t loadfnc); //called to register additional sound input plugins
wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength);
void S_AmbientOff (void);
void S_AmbientOn (void);
//inititalisation functions.
typedef int (*sounddriver) (soundcardinfo_t *sc, int cardnum);
extern sounddriver pOPENAL_InitCard;
extern sounddriver pDSOUND_InitCard;
extern sounddriver pALSA_InitCard;
extern sounddriver pSNDIO_InitCard;
extern sounddriver pOSS_InitCard;
extern sounddriver pSDL_InitCard;
extern sounddriver pWAV_InitCard;
extern sounddriver pAHI_InitCard;
struct soundcardinfo_s { //windows has one defined AFTER directsound
char name[256]; //a description of the card.
struct soundcardinfo_s *next;
//speaker orientations for spacialisation.
float dist[MAXSOUNDCHANNELS];
vec3_t speakerdir[MAXSOUNDCHANNELS];
//info on which sound effects are playing
channel_t channel[MAX_CHANNELS];
int total_chans;
//mixer
volatile dma_t sn; //why is this volatile?
qboolean inactive_sound; //continue mixing for this card even when the window isn't active.
qboolean selfpainting; //allow the sound code to call the right functions when it feels the need (not properly supported).
int paintedtime; //used in the mixer as last-written pos (in frames)
int oldsamplepos; //this is used to track buffer wraps
int buffers; //used to keep track of how many buffer wraps for consistant sound
int samplequeue; //this is the number of samples the device can enqueue. if set, DMAPos returns the write point (rather than hardware read point) (in samplepairs).
//callbacks
void *(*Lock) (soundcardinfo_t *sc, unsigned int *startoffset); //grab a pointer to the hardware ringbuffer or whatever. startoffset is the starting offset. you can set it to 0 and bump the start offset if you need.
void (*Unlock) (soundcardinfo_t *sc, void *buffer); //release the hardware ringbuffer memory
void (*Submit) (soundcardinfo_t *sc, int start, int end); //if the ringbuffer is emulated, this is where you should push it to the device.
void (*Shutdown) (soundcardinfo_t *sc); //kill the device
unsigned int (*GetDMAPos) (soundcardinfo_t *sc); //get the current point that the hardware is reading from (the return value should not wrap, at least not very often)
void (*SetWaterDistortion) (soundcardinfo_t *sc, qboolean underwater); //if you have eax enabled, change the environment. fixme. generally this is a stub. optional.
void (*Restore) (soundcardinfo_t *sc); //called before lock/unlock/lock/unlock/submit. optional
void (*ChannelUpdate) (soundcardinfo_t *sc, channel_t *channel, unsigned int schanged); //properties of a sound effect changed. this is to notify hardware mixers. optional.
//driver-specific - if you need more stuff, you should just shove it in the handle pointer
void *thread;
void *handle;
int snd_sent;
int snd_completed;
int audio_fd;
// no clue how else to handle this yet!
#ifdef AVAIL_OPENAL
int openal;
#endif
};
extern soundcardinfo_t *sndcardinfo;
typedef struct
{
void *(*Init) (int samplerate); /*create a new context*/
void (*Start) (void *ctx); /*begin grabbing new data, old data is potentially flushed*/
unsigned int (*Update) (void *ctx, unsigned char *buffer, unsigned int minbytes, unsigned int maxbytes); /*grab the data into a different buffer*/
void (*Stop) (void *ctx); /*stop grabbing new data, old data may remain*/
void (*Shutdown) (void *ctx); /*destroy everything*/
} snd_capture_driver_t;
#endif